Attention recently has been directed to implementing a variety of communication services, including voice telephone service, over the worldwide packet data network now commonly known as the Internet.
In a simplified fashion the Internet may be viewed as a series of packet data switches or `routers` connected together with computers connected to the routers. The Information Providers (IPs) constitute the end systems which collect and market the information through their own servers. Access providers are companies such as UUNET, PSI, MCI and SPRINT which transport the information. Such companies market the usage of their networks.
FIG. 3 shows a simplified diagram of the Internet and various types of systems typically connected thereto. Generally speaking the Internet consists of Autonomous Systems (AS) type packet data networks which may be owned and operated by Internet Service Providers (ISPs) such as PSI, UUNET, MCI, SPRINT, etc. Three such AS/ISPs appear in FIG. 3 at 310, 312 and 314. The Autonomous Systems (ASs) are linked by Inter-AS Connections 311, 313 and 315. Information Providers (IPs) 316 and 318, such as America Online (AOL) and CompuServe, connect to the Internet via high speed lines 320 and 322, such as T1/T3 and the like. Information Providers generally do not have their own Internet based Autonomous Systems but have or use Dial-Up Networks such as SprintNet (X.25), DATAPAC and TYMNET.
By way of current illustration, MCI is both an ISP and an IP, SPRINT is an ISP, and the Microsoft Network (MSN) is an IP using UUNET as an ISP. Other information providers, such as universities, are indicated in exemplary fashion at 324 and are connected to the AS/ISPs via the same type connections here illustrated as T1 lines 326. Corporate Local Area Networks (LANs), such as those illustrated in 328 and 330, are connected through routers 332 and 334 and high speed data links such as T1 lines 336 and 338. Laptop computers 340 and 342 are representative of computers connected to the Internet via the public switched telephone network (PSTN) and are shown connected to the AS/ISPs via dial up links 344 and 346.
Recently, software has been developed that allows personal computer (PC) users to conduct two-way voice conversations over the Internet. An audio card in the PC digitizes speech inputs received via a microphone and converts digital speech signals received from the Internet into analog audio output signals. The software provides compression and decompression of the digital signals to permit voice communication at rates as low as 2800 bits/s. The two-way conversion between digital and analog and the compression and decompression together are generally identified as a `vocoder` functionality. The software also controls TCP/IP packet processing by the PC that is necessary to transmit and receive digital speech signals over the Internet. The PC typically accesses the Internet through a modem-to-modem call to the server or router of an Internet Service Provider (ISP).
U.S. Pat. No. 4,872,197, issued Apr. 21, 1987, to Dorsey et al., titled Verbal Computer Terminal System, describes a system for providing voice telephone access to computers. The system is one wherein remote computers of a conventional type may be addressed or accessed by multiple DTMF telephones and respond or provide output to such telephones in the form of speech derived from the data bases of the respective computers. The system includes means between the standard computer and the DTMF input and analog audio output for emulating computer terminals acceptable to the host computer, for example for converting text data from the computers into speech signals transmissible over telephone line to a caller.
Several providers of Internet telephony software now offer along with the software the use of Internet Phone Servers. These servers, usually in distant cities, are available for users of the software to choose in order to connect to the Internet Phone Network of the particular software provider. In a sense, the servers fulfill the function of a phone directory to access other Internet Phone users. When users of the software connect to the Internet Phone server they are provided with a list of other connected users. From this list a choice may be made and the user can make calls to the other connected parties. In addition to this telephone directory type listing, the connected users are also listed under sublists of topics of conversation. Thus the service is similar to the so called "chat rooms" that are available from ISPs for keyboard to keyboard communication. The obvious shortcoming of the service from a telephony standpoint is an inability to make a call to a telephone subscriber who may or may not own a computer or who may not be on line at the time that the calling party desires to establish a contact.
One system for providing such an Internet telephone service which overcomes this difficulty is described in Farris and Bartholomew U.S. patent application Ser. No. 08/634,543, filed Apr. 18, 1996, for Public Internet Protocol Transport Network. That application is owned by the assignee of the instant application and is incorporated by reference herein in its entirety.
According to the Farris and Bartholomew arrangement, a public switched telephone network utilizing program controlled switching systems controlled by common channel interoffice signaling (CCIS), and preferably an advanced intelligent network (AIN) CCIS network, is arranged in an architecture to provide a methodology for facilitating impromptu telephone customer use of the Internet. Provision is made to permit a caller to set-up and carry out a telephone call over the Internet from telephone station to telephone station without customer access to computer equipment and without the necessity of the customer maintaining a subscription to any Internet service. Billing may be accomplished on a per call basis. The calls may be inter and intra LATA, region or state and may be nationwide. Usage is made of CCIS signaling to set up the call and establish the necessary Internet connections and addressing. Calls may be made from telephone to telephone, from voice capable computer to voice capable computer, or from telephone to computer or computer to telephone.
Another system for providing Internet telephone service on a small business basis is described in Michael J. Strauss application Ser. No. 08/815,358, filed Mar. 11, 1997. That application is also assigned to the assignee of the instant application and is incorporated by reference herein in its entirety. According to that application a small business interexchange Internet telephone service is provided using a pair of personal computers or PCs at distal sites. Each computer is connected to a central office of a Local Exchange Carrier (LEC) by two lines at the respective sites. The computers are provided with speech cards and the software for performing the vocoder and TCP/IP communication functionalities. The computers also have software to allow DTMF digit dialing and collection on both lines.
In response to a call on one of the lines, the called computer answers the call and collects digits corresponding to a PIN number and a desired voice call destination. The computer then initiates a modem data call on the other line to any ISP. The computer executes a log-in procedure with the ISP's modem pool and then initiates an Internet data session with the other distal computer, which computer serves the region covering the destination telephone number. That computer communicates with its ISP on one of its two lines and establishes a local voice telephone call to the dialed destination on the other line. The two lines to each computer are connected or bridged in the respective computers to complete an Internet interexchange link between the calling and called telephones connected to the two distal LECs.
A more versatile system for providing universal and multi-purpose telecommunication network to internetwork service is described in Strauss and Farris application Ser. No. 08/789,809, filed Jan. 28, 1997. That application is assigned to the assignee of the current application and is incorporated by reference herein in its entirety. The Strauss and Farris application describes a multi-purpose or multi-mode network server. The server provides enhanced processing functions in association with a telecommunications network to provide multi-mode communications via a combination of the public switched telephone network (PSTN) and a public packet data network, such as the Internet.
The improved network server includes a multiplicity of application processing units optimized for the processing of specific signal types. The type of signals being handled by the server is ascertained by means of a passive monitor and each type of signal is switched to an application processor on the basis of the signal type identified by the passive monitor. The processor places the processed signal in the protocol of the public packet data network and delivers that signal to a router connected to that network. Provision is made for establishing the availability of a called party through a control network, such as a CCIS network, before establishing an end to end communication link.
In addition to the foregoing a number of other publications have dealt with various types of telephony in switched packet networks.
The book "Mastering the Internet", Glee Cady and Pat McGregor, SYBEX Inc., Alameda, Calif., 1994, ISBN 94-69309, very briefly describes three proprietary programs said to provide real-time video and voice communications via the Internet.
Palmer et al. U.S. Pat. No. 5,375,068, issued Dec. 20, 1994, for Video Teleconferencing for Networked Workstations, discloses a video teleconferencing system for networked workstations. A master process executing on a local processor formats and transmits digital packetized voice and video data, over a digital network using TCP/IP protocol, to remote terminals.
Lewen et al. U.S. Pat. No. 5,341,374, issued Aug. 23, 1994, for Communication Network Integrating Voice Data and Video with Distributed Call Processing, discloses a local area network with distributed call processing for voice, data and video. Real-time voice packets are transmitted over the network, for example to and from a PBX or central office.
Hemmady et al. U.S. Pat. No. 4,958,341, issued Sep. 18, 1990, for Integrated Packetized Voice and Data Switching System, discloses an integrated packetized voice and data switching system for a metropolitan area network (MAN). Voice signals are converted into packets and transmitted on the network. Tung et al. U.S. Pat. No. 5,434,913, issued Jul. 18, 1995, and U.S. Pat. No. 5,490,247, issued Feb. 6, 1996, for Video Subsystem for Computer Based Conferencing System, disclose an audio subsystem for computer-based conferencing. The system involves local audio compression and transmission of information over an ISDN network.
Hemmady et al. U.S. Pat. No. 4,872,160, issued Oct. 3, 1989, for Integrated Packetized Voice and Data Switching System, discloses an integrated packetized voice and data switching system for metropolitan area networks. Sampat et al. U.S. Pat. No. 5,493,568, issued Feb. 20, 1996, for Media Dependent Module Interface for Computer Based Conferencing System, discloses a media dependent module interface for computer based conferencing system. An interface connects the upper-level data link manager with the communications driver.
Koltzbach et al. U.S. Pat. No. 5,410,754, issued Apr. 25, 1995, for Bi-Directional Wire Line to Local Area Network Interface and Method, discloses a bi-directional wire-line to local area network interface. The system incorporates means for packet switching and for using the internet protocol (IP).
These recent developments have lead to wider consideration of ways to make long distance and like calls through the Internet, for example to bypass interexchange (long distance) telephone carriers. As a result, telephone servers have been proposed which would provide interfaces for people using only standard telephones. The servers may receive and initiate telephone calls and perform either a vocoder functionality or a transcoding functionality (between digital telephone network encoding and the appropriate encoding for transport over the Internet). An in-bound server would receive an incoming call and collect destination information from the caller. That server would communicate via the Internet with a distant server. The distant server would perform out-bound functions, such as dialing, to establish a local call to the destination station. The two servers would then set up a voice communication link through the Internet.
While the foregoing types of service are effective to accomplish telephony service over the Internet, the actual call set-up, generally speaking, is time consuming and in some instances cumbersome. For example, according to one procedure a caller may have to dial a number, connect with a computer, dial a PIN (Personal Identification Number), engage in an IVR (Interactive Voice Response) dialog, and dial yet additional numbers to identify the destination desired. According to yet another mode used for connection the caller must serially dial an inordinate number of digits.